Well... the next step (for me anyway) would be to use Ethereal on the
asterisk nic interface to ensure the sip/rtp pkts are reasonable (eg, no
dropouts). If those pkts flow consistently in both directions, then
there must be something impacting the wctdm interface.
Do sip to sip calls sound reasonable?
Is there anything else running on your asterisk box?
sdgesa gaeharth wrote:
thanks for the info.
it is not sharing an irq:
0: 59840409 59803082 IO-APIC-edge timer
8: 1 0 IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
11: 0 0 IO-APIC-level ohci_hcd:usb1
14: 2141851 2143209 IO-APIC-edge ide0
177: 111558 111273 IO-APIC-level aic7xxx
185: 15 0 IO-APIC-level aic7xxx
193: 736328 748953 IO-APIC-level eth0
201: 239290099 239259220 IO-APIC-level wctdm
NMI: 0 0
LOC: 119645889 119645888
ERR: 0
MIS: 0
I checked the switch. The net connection is running at full duplex:
FastEthernet0/15 is up, line protocol is up (connected)
Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f)
MTU 1500 bytes, BW 100000 Kbit, DLY 100 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulatio n ARPA, loopback not set
Keepalive set (10 sec)
Full-duplex, 100Mb/s, media type is 100BaseTX
input flow-control is unsupported output flow-control is unsupported
ARP type: ARPA, ARP Timeout 04:00:00
Last input never, output 00:00:03, output hang never
Last clearing of "show interface" counters never
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 31000 bits/sec, 15 packets/sec
5 minute output rate 32000 bits/sec, 15 packets/sec
679924 packets input, 225898296 bytes, 0 no buffer
Received 3803 broadcasts (0 multicast)
0 runts, 0 giants, 0 throttles
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 watchdog, 5 multicast, 0 pause input
0 input packets with dribble condition detected
689110 packets output, 145860377 bytes, 0 underruns
0 output errors, 0 collisions, 2 interface resets
0 babbles, 0 late collision, 0 deferred
0 lost carrier, 0 no carrier, 0 PAUSE output
0 output buffer failures, 0 output buffers swapped out
*/Rich Adamson <[EMAIL PROTECTED]>/* wrote:
Based only on what I see below (from previous posts), it sounds like
you
have two separate issues going on: 1) echo, and, 2) choppy sound. Those
should be analyzed as two problems (not one).
You will find plenty of posts in the archives relative to both. In
general terms, the choppy audio m ost often is caused by shared IRQ's
when using a x100p or TDM400 card, and sometimes from a misconfigured
ethernet nic on the asterisk machine. For the nic card, ensure you are
running full duplex on the nic "and" whatever the nic is plugged in to.
Both need to be the same (half duplex will work in a low usage
environment, but full duplex is preferred.)
For the IRQ issue (and we are all assuming you are using a TDM04b card
since you really didn't say), do a 'cat /proc/interrupts' and make sure
your TDM card is on its own IRQ. If it is shared with other devices, it
is likely the cause for choppy audio. You'll see the TDM driver
wctdm on
that list. If it is shared, then move the TDM card from one pci slot to
another to get it on its own IRQ.
The echo problem is going to be almost aways related to "too high" of
gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as
others have already noted. Try reducing those to 0 and restart asterisk.
Then increase the values (if needed) by increments of 2 until you
find a
balance between low volume and echo. I'd suggest doing that "after"
resolving the IRQ/choppy audio issues.
> I have done this but I still get choppy sound and echo on some calls
>
> thanks
>
> */Giovanni Miano /* wrote:
>
> Of course,
> Echo is 2 types: electric and ambiental.
>
> If u gain rx o tx more than you need, its return in recive and
gen echo
>
> Try to decrase value, try to set 0 or .. in samecase -1 -2...
>
> 2006/3/13, sdgesa gaeharth
> >:
>
> Can you explain why?
>
>
> */Giovanni Miano
> >/* wrote:
>
&g t; rxgain=10.0
> txgain=10.0
>
> ????
>
> Maybe this is a problem
>
>
> 2006/3/13, sdgesa gaeharth < [EMAIL PROTECTED]
> >:
>
> I still hear a slight echo of my voice when I talk with
> somone out the PSTN. The voice on the other end sounds
> very choppy and a little distorted. When I talk to other
> people within our office, the sound is perfect.Can some
> help?
>
> We are all using:
>
> Polycom 501 <--> asterisk <--> PSTN
>
> zapata.conf:
> [channels]
> group = 1
> language=en
> context=incoming
& gt; signalling=fxs_ks
> switchtype=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> musiconhold=default
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echotraining=yes
> echocancelwhenbridged=yes
> rxgain=10.0
> txgain=10.0
> channel => 1-4
>
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